similar to: Testing

Hi all! I ran across another song that oggenc seems to have trouble to encode. The intro of U2:s "Grace" gets its right channel messed up by oggenc, even at the highest bitrate. Why is the bitrate only 218? Isn't that awfully little when asking for 320? Qualityflaws like these is the reason i prefer mp3 and mpplus today. But vorbis is getting better for every beta, so keep up the

Hi! My winamp crashes when i play this file: http://rehlin.hemmet.chalmers.se/~papa/files/test.ogg the wav is in the same place. Does someone else have the same 'problem'? (ok, i know its a silly test sound, but i still think it should decode without crashing) ome numbers: betaversion of oggenc: 4 bitrate used: 350 version of winamp: 2.74 version of vorbis plugin: 1.04 windows: 2000 pro

After reading http://mp3.radified.com/mp3.htm I sent Rad an explanation of some things as I understand them. He liked it and posted it on his site (still unlinked, use the URL below). Can somebody with better understanding of psycho-acoustic terms and the vorbis model check it and comment on it? In particular I didn't know how vorbis handles quantization noise. If you reply with

Hi! I am very pleased with the progress that Ogg is making, expecially after I read the latest comparision tests on http://ff123.net/128test/instruct.html that put OGG on top aside with MPC. BUT the APPLAUD.WAV test case ( http://lame.sourceforge.net/download/samples/applaud.wav ) still produces **VERY** easily audible high-frequency artifacts when encoded with OGG RC3 up to q4.9 (!!!). Things

http://bugzilla.mindrot.org/show_bug.cgi?id=52 ------- Comment #47 from tsi at ualberta.ca 2007-02-17 03:06 ------- (In reply to comment #46) > Yes, but that workaround doesn't need the patch. However, with latest > patch, there is no hang when I run test2.ksh and exit from the shell on > linux. But it is not the case with HP-UX. So Im wondering what the > patch is doing with

The Vorbis audio format, uses something called the Modified Discrete Cosine Transform (MDCT). I was interested in how this worked, but did not know where to start. Is this the name of a particular algorithm, or does it just mean something like a cosine transform, but not quite right? So I asked the Vorbis mailing list. Nobody answered. It was not a high priority, so I did other things for

Hello: Does anyone know if Garf's tunings have been integrated into CVS yet? I am waiting for that before upgrading. Charles -- "Problem solving under linux has never been the circus that it is under AIX." (By Pete Ehlke in comp.unix.aix) -------------- next part -------------- A non-text attachment was scrubbed... Name: part Type: application/pgp-signature Size: 190 bytes

My icecast server (1.3.11) has been getting 404 errors for the past couple of days from the yp.shoutcast.com server. If I send the URL manually by telneting to the webserver at yp.shoutcast.com I get the following response: [chrisp@ayeka httpd]$ telnet yp.shoutcast.com 80 Trying 205.188.234.56... Connected to scastlb1.shoutcast.com. Escape character is '^]'. GET

Hello. First of all, thanx for that great codec. I would encode all my music with Ogg Vorbis in the future, if it hadn't been for some samples I encoded for test purposes: Everything was great as long a I used "natural" sounds/music, but certain pieces of synthesized music (specially techno) contained very audible artifacts even at high bitrates. So I just went on using mp3 for

Hi, Was wondering if any gurus could tell me if it is possible to stream Ogg on very low bandwidth links (for example satellite phone) where the expected bandwidth would almost certainly be less than 19Kbps, and probably more like 10Kbps. I am not after great audio at this bitrate, mono with an audio bandwidth less than 5KHz. I have done some testing (not streaming though) listening to 8Kbps MP3

Hello, I'm seeking advice here as I'm intend in the not to distant future to start encoding my CD collection, and I'm not sure what to use. Do I use the still current v1.0, or am I better off using a newer build? Also, I've started to notice mention of versions 1.0.1 and 1.1 in these mailing lists. Are these worth holding out for? I'll probably encode at about Q7, and am using

Hello All, Just a short note to add my vote for a HUGE round of applause to the CentOS team for their untiring efforts in getting releases out the door. I've just upgraded several servers to 5.6 and it all "just works". None of the team's work is easy to accomplish, especially when less-than-useful complaints keep popping up from thoughtless users who don't appreciate

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

Ok, ok I am a "one minute" perl expert now (sung to the tune of "I read the One Minute Manager Book today yeah yeah yeah baby baby" :p My first perl research and program called test.pl use Net::DNS; print Net::DNS->version, "\n"; the result of "perl test.pl" as expected was 0.48 Please keep the applause to a minimum please. Tens and twenties work

Hey, last patch for screen.css: --- screen.css.orig 2007-11-21 17:18:13.000000000 +0000 +++ screen.css 2007-11-21 17:22:43.000000000 +0000 @@ -13,8 +13,6 @@ */ body { - padding-top: 3%; - padding-bottom: 3%; padding-right: 10%; padding-left: 10%; margin: 0px; And after that we go live! Just joking. We already are live. See <http://wiki.centos.org/>. And

Hi! I got my first Samba 4 system up and running. {No applause, just throw money. Paper please, coins hurt} It is just a home network of about a dozen or so boxes. I have noticed that when I do a testparm, the result is not the same as the file I created. I'm fairly certain it isn't anything to worry about; I'm just trying to learn. Thanks! -jimc

hello there, I just started using ogg files. And want to thank you all for this new format. I have written an addon for mirc (chatclient for irc chat networks) to play and exchange soundfiles and I am almost done with implementing the ogg vorbis format into this application. The mirc-scriptinglanguage provides me a command to read a specified number of bytes starting at a specified location of

Hi All, I've run into a small hiccup in encoding my audios with Speex. When I encode audience laughter and applause with 'speexenc' (version 1.0.1), the result is quite acceptable... until I enable VBR. Then it distorts horribly. My understanding of VBR is that it frees the encoder to vary the number of bits emitted to better maintain the quality requested, and so I would have

I took 5 albums (Classical music) and converted them to Ogg Vorbis at "Full Bitrate" (-q10) and all 5 directories take up about 775 Megs which won't fit on a CD. So I ripped them again in WAV first (And give my friend back his CDs) but now I wanna know what quality setting should I use to fit them on 1 CD (The highest possible with total space used just under 700 Megs)

Hi, In paper" A High-Quality Speech and Audio Codec With Less Than 10 ms Delay", you are mention that: There are still several ways to improve CELT, such as by incorporating better psychoacoustic masking in the dynamic bit allocation. This is a difficult problem both because there are few bits available for coding the allocation and because the analysis window is short. it is